Systems and methods for altering the input dynamic range of an auditory device

ABSTRACT

An auditory device according to one embodiment includes a processing unit configurable to determine a noise estimate and a measure for one of a plurality of samples of an audio signal. The measure relates to an upper volume bounds. The processing unit is configured to apply at least one rule using the noise estimate and/or the measure to identify an input dynamic range for mapping the audio signal to a corresponding stimulation signal.

BACKGROUND

Various types of auditory devices provide persons with different typesof hearing loss with the ability to perceive sound or perceive improvedsound, or persons having normal hearing to experience different soundperception (e.g. with less noise). Hearing loss may be conductive,sensorineural, or some combination of both conductive and sensorineural.Conductive hearing loss typically results from a dysfunction in any ofthe mechanisms that ordinarily conduct sound waves through the outerear, the eardrum, or the bones of the middle ear. Sensorineural hearingloss typically results from a dysfunction in the inner ear, includingthe cochlea, where sound vibrations are converted into neural signals,or any other part of the ear, auditory nerve, or brain that may processor convey the neural signals.

Persons with some forms of conductive hearing loss may benefit fromauditory devices such as acoustic hearing aids or vibration-basedauditory devices. An acoustic hearing aid typically includes a smallmicrophone to detect sound, an amplifier to amplify certain portions ofthe detected sound, and a small speaker to transmit the amplified soundsinto the person's ear. Vibration-based auditory devices typicallyinclude a small microphone to detect sound and a vibration mechanism toapply vibrations corresponding to the detected sound directly orindirectly to a person's bone or teeth, for example, thereby causingvibrations in the person's inner ear and bypassing the person's auditorycanal and middle ear. Vibration-based auditory devices include, forexample, bone-anchored devices, direct acoustic cochlear stimulationdevices, or other vibration-based devices. A bone-anchored devicetypically utilizes a surgically implanted mechanism or a passiveconnection through the skin or teeth to transmit vibrationscorresponding to sound via the skull. A direct acoustic cochlearstimulation device also typically utilizes a surgically implantedmechanism to transmit vibrations corresponding to sound, but bypassesthe skull and more directly stimulates the inner ear. Other non-surgicalvibration-based auditory devices may use similar vibration mechanisms totransmit sound via direct or indirect vibration of teeth or othercranial or facial bones or structures.

Persons with severe to profound sensorineural hearing loss may benefitfrom surgically implanted auditory devices (prostheses), such ascochlear implants, auditory brainstem implants, or auditory midbrainimplants. For example, a cochlear implant can provide a person havingsensorineural hearing loss with the ability to perceive sound bystimulating the person's auditory nerve via an array of electrodesimplanted within the cochlea. A component of the cochlear implantdetects sound waves, which are converted into a series of electricalstimulation signals that are delivered to the person's cochlea via thearray of electrodes. Auditory brainstem implants can use technologysimilar to cochlear implants, but instead of applying electricalstimulation to a person's cochlea, auditory brainstem implants applyelectrical stimulation directly to a person's brainstem, bypassing thecochlea altogether. Electrically stimulating auditory nerves in acochlea with a cochlear implant or electrically stimulating a brainstemmay enable persons with sensorineural hearing loss to perceive sound.

Further, some persons may benefit from auditory devices that combine oneor more characteristics of acoustic hearing aids, vibration-basedauditory devices, cochlear implants, and auditory brainstem implants toenable the person to perceive sound. Such auditory devices can bereferred to generally as hybrid auditory devices.

Common among some or all the above-described auditory devices is theneed to determine the stimulus to provide to the auditory devices'stimulation mechanism/device (electrode, vibrator, speaker, etc.), sothat the user of the auditory device is able to hear important sounds(information) at a loudness that is perceptible, yet comfortable for theuser. This requires, first, that the auditory device be properly fit tothe user, so that one or more stimulation channels, for example, provideappropriate maximum and minimum levels of stimulation to the user. Forexample, an acoustic hearing aid should be fit so that the hearing aid'sspeaker preferably does not cause discomfort to the user in the presenceof loud ambient sounds, but still allows the user to hear quiet ambientsounds. For electrical, rather than acoustic stimulation, fittingtypically refers to choosing an acceptable range of current levels to beprovided to one or more stimulation electrodes, or a stimulus signal tobe provided to a vibration mechanism or other source of stimulation.

In addition to fitting, how the auditory device initially handles theincoming ambient sounds may also be important. Noise reductiontechniques are commonly employed in auditory devices to attenuate partsof the signal that are determined to be noise, while retaining thetarget information content of the signal. Compression and expansiontechniques are also commonly employed in auditory devices to amplify orattenuate signals that are too soft or too loud to improve listeningcomfort and speech understanding.

While noise reduction can decrease the loudness of masking noise, it canalso, to a lesser extent, decrease the loudness of the target talker. Italso has no inherent limits, so in very noisy environments typical noisereduction schemes remove much of the noise signal, which reduceslistening quality. While compression and expansion systems can improvelistening comfort in loud or soft environments, they can also havenegative effects on listening quality. For example, in the case ofcompressors that attenuate short, loud sounds, they may also attenuatethe background during and after the short loud sound creating theperception of a transient pumping noise, which reduces listeningquality.

Thus, even if an auditory device is properly fit to its user andincludes some form of noise reduction and compression and expansionalgorithms, listening in noisy or loud or soft environments can resultin relatively poorer speech quality and intelligibility when listeningin real-world environments.

SUMMARY

Aspects of the present disclosure relate generally to sound processingand, more particularly, to improving the listening experience for a userof an auditory device. Generally, many typical auditory devices operatewith a fixed input dynamic range and utilize complicated multi-stagesignal paths, making it difficult to predict system effects caused bychanging a single amplitude-modifying element. Furthermore, thecombination of multiple algorithms, each modifying gains applied to theincoming signal, may behave in an unpredictable manner when operatingtogether in unison. The result may be a poorer performance or soundquality under some real-world listening environments.

The present disclosure relates to one or more aspects intended toimprove on prior multi-stage processing strategies. Generally, anauditory device incorporating aspects set forth herein includes aprocessor to adjust an input dynamic range based on a noise estimateand/or a measure, such as a percentile estimate of the input audiosignal. For example, the percentile estimate could represent theninetieth percentile input signal amplitude over a preceding timeperiod, such as thirty seconds. The processor uses the noise estimateand/or measure, which may be in units of decibels (dB) of Sound PressureLevel (SPL), for example, to set the input dynamic range. In this way,the input dynamic range adjusts to fit the listening environment. Theadjustments are preferably smoothed over a suitable time period, such asthirty seconds, to allow the user of the device to adapt. One or morerules may be applied to ensure that the dynamic range is not set toobroad or narrow, or too low or high. In addition, one or more breakrules may be applied to allow the device to quickly respond to soundslouder than a particular amplitude, such as those at a certainpercentage above the measure. The user's own voice, often louder thanthe far field target information, might trigger such a break rule, forexample. The result is an improved listening experience for the user,where important information, such as a target speaker's voice, is morereadily heard, while noise is advantageously suppressed.

More specifically, in accordance with one embodiment, an auditory deviceincludes a processing unit configurable to determine a noise estimateand a measure for one of a plurality of samples of an audio signal, suchas an input audio signal. The processing unit applies at least one ruleusing the noise estimate and/or the measure to identify an input dynamicrange for mapping the input audio signal to a corresponding stimulationsignal. The auditory device may further include a front end configurableto generate the plurality of samples and a stimulation unit forproviding a stimulus based on the stimulation signal to a user of theauditory device. The stimulus may be an electrical stimulus or amechanical stimulus, for example.

In a further embodiment, the processing unit is further configurable todetermine respective noise estimates and measure for other of theplurality of samples and to temporally smooth a plurality of the noiseestimates and measures while identifying the input dynamic range.Alternatively, the processing unit is further configurable to determinerespective noise estimates and measures for other of the plurality ofsamples, to identify a plurality of input dynamic ranges based on thedetermined respective noise estimates and measures, and to temporallysmooth the identified plurality of input dynamic ranges for mapping theaudio signal to a corresponding stimulation signal.

The at least one rule may include an information rule, a noise rule, orboth, for example. The information rule includes determining whether anupper value of the input dynamic range is greater than an upper valuemaximum limit and whether the upper value of the input dynamic range isless than an upper value minimum limit. Upon determining that the uppervalue of the input dynamic range is greater than the upper value maximumlimit or less than the upper value minimum limit, the information ruleincludes adjusts the upper value respectively to the upper value maximumlimit or the upper value minimum limit.

The noise rule includes determining whether a lower value of the inputdynamic range is less than a lower value minimum limit and whether thelower value of the input dynamic range is greater than a lower valuemaximum limit. Upon determining that the lower value of the inputdynamic range is less than the lower value minimum limit or greater thanthe lower value maximum limit, the noise rule calls for adjusting thelower value respectively to the lower value minimum limit or the lowervalue maximum limit.

The at least one rule to be applied could additionally or alternativelyinclude a stretch rule, a squash rule, or both. The stretch ruleincludes determining whether a difference between the upper value andthe lower value is greater than a maximum difference threshold. If so,the stretch rule adjusts at least one of the upper value and the lowervalue so that the difference is not greater than the maximum differencethreshold.

The squash rule includes determining whether the difference between theupper value and the lower value is less than a minimum differencethreshold. If so, the squash rule adjusts at least one of the uppervalue and the lower value so that the difference is not less than theminimum difference threshold.

The auditory device can further include at least one user input to setat least one of a volume and a sensitivity. The volume is used indetermining an upper value of the input or output dynamic range, whilethe sensitivity is used in determining a lower value of the input oroutput dynamic range.

According to this first embodiment, the processing unit maps the inputaudio signal to the corresponding stimulation signal according to a mapusing the lower value of the input dynamic range and an upper value ofthe input dynamic range. The input dynamic range maps to a correspondingoutput range predetermined for the user and the stimulation device. Forelectrical stimulation, the mapping may include determining a signalamplitude, a pulse width, or an inter-phase gap for the stimulationsignal, or some combination of these, for example. For acoustic orvibratory stimulation, the mapping may include a speaker volume orvibration rate or amplitude, for example.

The at least one rule could further comprise a break rule, in which theprocessor could additionally determine whether a magnitude of the inputaudio signal exceeds an upper value of the input dynamic range by abreak threshold. If it does, the processor could increase the uppervalue of the input dynamic range.

In accordance with a second embodiment, a method for processing an audiosignal, such as an input audio signal, comprises determining anamplitude data, setting an input dynamic CSPL (comfort sound pressurelevel) parameter based on the amplitude data, setting an input dynamicTSPL (threshold sound pressure level) parameter, and mapping the audiosignal to a stimulation signal corresponding to the audio signal, wherethe mapping is, at least in part, according to the input dynamic CSPLparameter and the input dynamic TSPL parameter. In one example, theamplitude data is a measure of a characteristic of an average peakamplitude of the audio signal. The method may further include providinga stimulation according to the stimulation signal.

The method may further include performing at least one of the followingdeterminations and adjustments: (a) determining whether the inputdynamic CSPL parameter is greater than an upper value maximum limit, andif so, adjusting the input dynamic CSPL parameter to no more than theupper value maximum limit; (b) determining whether the input dynamicCSPL parameter is less than an upper value minimum limit, and if so,adjusting the input dynamic CSPL parameter to no less than the uppervalue minimum limit; (c) determining whether the input dynamic TSPLparameter is greater than a lower value maximum limit, and if so,adjusting the input dynamic TSPL parameter to no more than the lowervalue maximum limit; (d) determining whether the input dynamic TSPLparameter is less than a lower value minimum limit, and if so, adjustingthe input dynamic TSPL parameter to no less than the lower value minimumlimit; (e) determining whether a difference between the input dynamicCSPL parameter and the input dynamic TSPL parameter is greater than amaximum difference threshold, and if so, adjusting at least one of theinput dynamic CSPL parameter and the input dynamic TSPL parameter sothat the difference is not greater than the maximum differencethreshold; and (f) determining whether the difference between the inputdynamic CSPL parameter and the input dynamic TSPL parameter is less thana minimum difference threshold, and if so, adjusting at least one of theinput dynamic CSPL parameter and the input dynamic TSPL parameter sothat the difference is not less than the minimum difference threshold.

The method may yet further include determining whether a magnitude ofthe audio signal exceeds the input dynamic CSPL parameter by a breakthreshold, and if so, increasing the input dynamic CSPL parameter.

The stimulation signal may, for example, be an electrical stimulationsignal characterized by a C-Level current level and a T-Level currentlevel, where the input dynamic CSPL parameter is mapped to the C-Levelcurrent level, the input dynamic TSPL parameter is mapped to the T-Levelcurrent level, and both the C-Level current level and the T-Levelcurrent level are predetermined. The mapping may include determining asignal amplitude, a pulse width, and/or an interphase gap for thestimulation. Alternatively or additionally, the mapping may includedetermining a stimulus level according to one or more of a stimuluslevel, a current level, a pulse width, a vibration rate, and aninterphase gap.

Setting the input dynamic TSPL parameter may further include determininga noise-level estimate during a noise smoothing time period, and settingthe input dynamic TSPL parameter to the noise-level estimate.

A method according to a third embodiment includes determining a measureand a noise estimate for an audio signal, adjusting an input dynamicrange based on the measure and the noise estimate, and determining anoutput representative of a stimulus level based on the measure, thenoise estimate, and a static input. In one example, the audio signal isan input signal composed of a plurality of samples, and the adjusting isperformed for one of the plurality of samples. In another example, theaudio signal is a channelized input audio signal and the method includesdetermining a measure and a noise estimate, adjusting an input dynamicrange, and determining an output for each channel in the channelizedsignal. Determining the output may additionally include applying atleast one user input corresponding to volume or sensitivity, where theuser inputs are received asynchronously. Alternatively or additionally,the static input includes a device-specific parameter and/or auser-specific parameter. As other examples, the static input includes aminimum sound pressure level corresponding to a noise floor, a maximumsound pressure level corresponding to a maximum value of ananalog-to-digital conversion, and a T-Level and C-Level correspondingrespectively to a minimum and maximum of an electrical dynamic range ofan acoustic-to-electric mapping function determined using at least themeasure, the noise estimate, and/or the static input. The method furtherpreferably includes adjusting an acoustic-to-electric mapping functionbased on the measure and the noise estimate, where determining theoutput includes applying the acoustic-to-electric mapping function tothe audio signal. The method may further include communicating theoutput to a stimulation device, such as by applying anacoustic-to-electric mapping function to the input signal.

Yet another embodiment provides a method for processing an audio signalin an auditory device having first processing unit associate with afirst device and a second processing unit associated with a seconddevice. The auditory device may be a bilateral auditory device, forexample. The method includes determining, at the first processing unit,an input dynamic range based, at least in part, on a communicationreceived from the second processing unit. The communication isassociated with the second device and includes a dynamic TSPL, a dynamicsmoothed TSPL, a dynamic CSPL, a dynamic smoothed CSPL, a break ruleoutput, a scaling value input, a volume input, a sensitivity input,and/or a Q-Value input. The method further includes determining, at thefirst processing unit, an output representative of a stimulus level forthe first device, based on the determined input dynamic range. Themethod may further include the first processing unit communicating thedetermined input dynamic range to the second processing unit.

The above and additional aspects, examples, and embodiments are furtherdescribed in the present disclosure.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1A is a schematic block diagram illustrating an auditory device.

FIG. 1B is a schematic diagram illustrating an electrode array for theauditory device of FIG. 1A.

FIG. 1C is a conceptual illustration of a dynamic range curve mappingbetween acoustic signal loudness and electrode array current level (dB).

FIG. 2 is a block diagram of a processing unit depicted in FIG. 1A,according to an example.

FIG. 3 is a block diagram of an implanted unit depicted in FIG. 1A,according to an example.

FIG. 4 is a functional block diagram illustrating a model of inputs andoutputs that may be included in various embodiments.

FIG. 5 is a functional block diagram illustrating a device fordynamically adjusting an input dynamic range and mapping the inputsignal to a new output signal, in accordance with embodiments of thepresent invention.

FIG. 6 is a functional block diagram illustrating the rules module ofthe device of FIG. 5, according to an example.

FIG. 7 is a graphical representation of an acoustic-to-electric mappingcurve, according to an example.

FIG. 8 is a graphical representation of an acoustic-to-acoustic mappingcurve illustrating effects of adjustments to additional user inputs forthe mapping function module, according to an example.

FIG. 9 is a flow diagram illustrating a method according to a firstmethod.

FIG. 10 is a flow diagram illustrating a method according to a secondmethod.

DETAILED DESCRIPTION

The following detailed description describes various features,functions, and attributes with reference to the accompanying figures. Inthe figures, similar symbols typically identify similar components,unless context dictates otherwise. The illustrative embodimentsdescribed herein are not meant to be limiting. Certain features,functions, and attributes disclosed herein can be arranged and combinedin a variety of different configurations, all of which are contemplatedin the present disclosure.

Various embodiments of the present disclosure may be implemented inconjunction with a variety of auditory devices commercially availabletoday or as may be developed in the future. For example, the auditorydevices may include prosthetic hearing devices, as well as acousticdevices for the outer, middle, and inner ear, and others. Thus, cochlearimplants, including totally implantable cochlear implants, boneconduction devices, direct acoustic stimulation devices, auditory brainstem implants, middle ear implants, and/or other auditory devices arecontemplated by the present disclosure. Further, many features andfunctions disclosed herein may be equally applicable to devices otherthan prosthetic hearing devices, including other types of medical andnon-medical devices. In addition, various embodiments of the presentdisclosure are applicable to both unilateral and bilateral auditorydevices (and others). However, for ease of illustration, embodiments aredescribed herein in conjunction with auditory devices, and inparticular, with a unilateral cochlear implant device.

Generally, embodiments described herein are directed to altering aninput dynamic range (i.e. dynamically changing the CSPL and/or TSPL)during operation or use of an auditory device to help ensure that themost important part of an input audio signal is used for stimulation. Ata high level, this dynamic CSPL and dynamic TSPL concept is achieved bymapping an upper portion (e.g. a 95^(th) percentile estimate) of theinput audio signal to a C-Level (upper portion of the dynamic rangeperceived as loud) and by mapping a lower portion (e.g. a noiseestimate) to a T-Level (lower portion of the dynamic range perceived assoft).

FIG. 1A is a schematic block diagram illustrating an auditory device 100in the form of a cochlear implant. The auditory device 100 includes aprocessing unit 110 and an implanted unit 112. The implanted unit 112 isimplanted in a portion of a skull of a user under their skin 116. In anexample in which the auditory device 100 is a totally implantablehearing prosthesis, the processing unit 110 is also implanted in theuser's skull and below the user's skin 116. Additionally, an enclosuremay house the processing unit 110 and the implanted unit 112.

The implanted unit 112 includes an electrode array 114 implanted in theuser's cochlea. Each electrode on the electrode array 114 stimulates aportion of the cochlea that allows the user to perceive sound having arange of frequencies. The electrodes in the electrode array 114 deliverelectrical stimuli to one or more portions of the user's cochlea toallow the user to perceive at least a portion of a sound.

FIG. 1B illustrates the electrode array 114, which includes electrodesL1-L8. The electrode array 114 may include more or fewer electrodes. Forinstance, the electrode array 114 may include twenty-two electrodes. Tofacilitate implantation, the electrode array 114 is made of a flexiblematerial suitable for being implanted in the user's cochlea. When anelectrode, such as electrode L8, delivers an electrical stimulus, theuser perceives a sound having a particular frequency (e.g. about 1500Hz).

The volume of the sound perceived by the user of the auditory device 100depends on the stimulus current of each electrode provided by theprocessing unit 112 upon receiving a sound. The stimulus currents arethe currents of the electrical stimuli delivered by one or more of theelectrodes L1-L8. In general, as the current of a stimulus increases,the loudness of the sound perceived by the user increases. A value of astimulus current depends on a sound pressure level (SPL) of a soundreceived by the processing unit 110.

FIG. 1C is a conceptual illustration of a dynamic range curve 150 of anelectrode on the electrode array 114. The dynamic range curve 150 isplotted on a log-linear scale, with the x-axis representing a logarithmof the SPL of a sound, and the y-axis representing the stimulus currentin linear units. The stimulus current is expressed in any unit suitablefor use in the auditory device 100. In one example, the unit ismicroamperes. In another example, the unit is any unit capable of beingconverted to microamperes.

The dynamic range for the electrical stimuli from an electrode is adifference between a threshold level (T-Level) and a maximum comfortlevel (C-Level). A T-Level for an electrode corresponds to a stimuluscurrent that results in the user just being able to hear a sound at agiven frequency. In other words, the T-Level is typically the stimuluscurrent below which the user is not able to perceive the sound. TheC-Level for an electrode typically corresponds to the stimulus currentapplied by the electrode to the user's cochlea that results in a soundwith a certain pitch percept that the user can perceive comfortably.That is, the C-Level is the stimulus current above which the userperceives the sound as being uncomfortably loud.

The SPL of the sound at the acoustic threshold level is identified asthe output TSPL, and the SPL of the sound at the acoustic comfort levelis identified as output CSPL. Output TSPL represents the SPL of thesound below which amplification is needed to allow the user to perceivethe sound. Similarly, output CSPL represents the SPL of the sound abovewhich the sound becomes uncomfortably loud. Between output TSPL andoutput CSPL, the log outputs increase approximately linearly with thelog-SPL inputs.

For a sound having an SPL that is less than the dynamic TSPL, the soundwill be mapped to below the T-Level, and not perceived. For a soundhaving an SPL that is between the dynamic TSPL and the dynamic CSPL, thestimulus current varies approximately linearly with the SPL (dB) of thesound. The slope of the dynamic range curve 150 may have some additionalcurvature due to the Q-Factor (see, e.g., description accompanying FIGS.4 and 5). In one example, the slope of the dynamic range curve 150 isadjustable, allowing the dynamic range curve 150 to be customized to theuser. For a sound having an SPL greater than the output CSPL (orC-Level), the stimulus current is typically fixed at the C-Level. Inother words, the output CSPL is the saturation level for the electrode.In accordance with various embodiments described herein, adaptiveadjustments are made to the dynamic TSPL and/or dynamic CSPL duringoperation of the auditory device.

Each of the electrodes L1-L8 has a dynamic range curve. While differentusers of hearing prostheses like the auditory device 100 may havesimilar types of hearing losses (e.g., sensorineural hearing loss), eachuser may have a unique sensitivity to sounds at different frequencies.To accommodate the user's specific hearing loss, the auditory device 100is fit to the user of the auditory device 100 using a computing devicehaving a wired or wireless connection. Fitting the auditory device 100to the user includes determining the dynamic range for each of theelectrodes L1-L8.

FIG. 2 is a block diagram of a processing unit 200 of an auditorydevice. The processing unit 200 is one example of the processing unit110 depicted in FIG. 1A. The processing unit 200 includes a power supply202, an audio transducer 204, a data storage 206, a sound processor 208,a transceiver 212, and an inductive coil 214, all of which may beconnected directly or indirectly via circuitry 220.

The power supply 202 supplies power to various components of theprocessing unit 200 and can be any suitable power supply, such as arechargeable or non-rechargeable battery. In one example, the powersupply 202 is a non-rechargeable battery that can be easily replacedwhen it becomes depleted. In another example, the power supply 202 is arechargeable battery and can be disconnected from the processor,recharged, and re-connected at a later date. In another example, thepower supply 202 is a non-replaceable rechargeable battery that isrecharged wirelessly such as through an inductive power transfer system.The power supply 202 also provides power to the implanted unit of theauditory device 100 via the inductive coil 214.

The audio transducer 204 receives a sound from an environment and sendsa sound signal to the sound processor 208. In one example, theprocessing unit 200 is part of a bone conduction device, and the audiotransducer 204 is an omnidirectional microphone. In another example, theprocessing unit 200 is part of a cochlear implant, an auditory brainstem implant, a direct acoustic stimulation device, a middle earimplant, or any other auditory device now known or later developed thatis suitable for assisting a user of the auditory device 100 inperceiving sound. In this second example, the audio transducer 204 is anomnidirectional microphone, a directional microphone, anelectro-mechanical transducer, or any other audio transducer now knownor later developed suitable for use in the type of auditory deviceemployed. Furthermore, in other examples the audio transducer 204includes one or more additional audio transducers.

The data storage 206 includes any type of non-transitory, tangible,computer readable media now known or later developed configurable tostore program code for execution by a component of the processing unit200 and/or other data associated with the processing unit 200. The datastorage 206 also stores information indicative of a dynamic range forthe electrodes L1-L8 of the electrode array 114. The data storage 206may also store software programs executable by the sound processor 208.

The sound processor 208 receives an audio signal and processes the audiosignal to determine a stimulation signal suitable for use by theauditory device, such as in an implanted unit associated with theauditory device. In one example, the sound processor 208 is a digitalsignal processor. In another example, the sound processor 208 is anyprocessor or microcontroller or combination of processors andmicrocontrollers now known or later developed suitable for use in anauditory device. Additionally, the sound processor 208 may includeadditional hardware for processing the audio signal, such as ananalog-to-digital converter and/or one or more filters.

The sound processor 208 is configured to receive the audio signal fromthe audio transducer 204 in the form of the sound signal. In processingthe audio signal, the sound processor 208 identifies an SPL of the audiosignal at one or more frequencies. The sound processor 208 accesses thedata storage 206 to identify information indicative of the dynamic rangefor one or more of the electrodes L1-L8 in order to generate thestimulation signal. Based on the SPL of the sound at a given frequency,the sound processor 208 determines which of the electrodes L1-L8 need tostimulate the user's cochlea to allow the user to perceive the sound.The sound processor 208 determines the current to be applied through oneor more of the electrodes L1-L8 to stimulate the user's cochlea based onthe electrodes' dynamic range, also known as maps.

The transceiver 212 receives the stimulation signal from the soundprocessor 208 and modulates the stimulation signal to form atransmission signal. The transmission signal also includes the powersignal received from the power supply 202. In one example, thetransceiver 212 modulates the stimulation signal using a time-divisionmultiple-access modulation scheme. In another example, the transceiver212 uses any modulation scheme now known or later developed suitable forinductively transmitting the stimulation signal to an implanted unit ofan auditory device. The transceiver 212 sends the transmission signal tothe inductive coil 214.

The inductive coil 214 receives the transmission signal from thetransceiver 212 and inductively transmits the transmission signal to theimplanted unit 112. The inductive coil 214 is constructed of anymaterial or combination of materials suitable for inductivelytransferring a power signal to the implanted unit.

FIG. 3 is a block diagram of an implanted unit 300 of an auditorydevice, such as a cochlear implant. The implanted unit 300 is oneexample of the implanted unit 112 depicted in FIG. 1A. The implantedunit 300 includes an inductive coil 302, power management 304, and astimulation decoder 306, all of which are connected directly orindirectly via circuitry 310. The implanted unit 300 also includes astimulation component 308 that is connected to the stimulation decoder306 via circuitry 312. The stimulation component 308 is an example of astimulation device.

The inductive coil 302 receives the transmission signal from theprocessing unit 110. The inductive coil 302 is constructed of anybiocompatible material or combination of materials suitable forinductively receiving power from the processing unit 200. The inductivecoil 302 transfers the power signal to the power management 304.Alternatively, the implanted unit 300 may not include the powermanagement 304. In this case, the inductive coil 302 transfers the powersignal to the stimulation decoder 306 and the stimulation component 308.

The power management 304 receives the transmission signal from theinductive coil 302 and distributes power to the components of theimplanted unit 300. The power management 304 also includes a componentsuitable for removing the coded stimulation signal from the powersignal. The power management 304 sends the coded stimulation signal tothe stimulation decoder 306. The stimulation decoder 306 decodes thecoded stimulation signal and transfers the stimulation signal to thestimulation component 308.

The stimulation component 308 receives the stimulation signal from thestimulation decoder 306 and generates a stimulus based on thestimulation signal. In one example, the stimulation component 308includes a first subcomponent configured to generate the stimulus and asecond subcomponent configured to deliver the stimulus to an auditoryorgan, such as a cochlea, an auditory nerve, a brain, or any other organor body part capable of assisting a user of the auditory device inperceiving at least a portion of a sound. The first subcomponentgenerates the stimulus based on the stimulation signal and sends thestimulus to the second component. The second subcomponent delivers thestimulus to the body part of the user.

For instance, since implanted unit 300 is part of a cochlear implant inthe illustrated example, the stimulation component 308 includes a signalgenerator and the electrode array 114. The signal generator generates anelectrical signal based on the stimulation signal and sends theelectrical signal to the electrode array 114. The electrical signalcauses one or more of the electrodes L1-L8 to deliver one or moreelectrical stimuli to a portion of the user's cochlea. The one or moreelectrical stimuli cause the cochlea to stimulate the user's auditorynerve, thereby allowing the user to perceive at least a portion of asound.

In another example, the stimulation component 308 stimulates a differentbody part of the user. For instance, if the auditory device is anauditory brain stem implant, the stimulation component 308 provides thestimulation signal directly to the user's brain. In this case, thestimulation component 308 includes an electrode array that is implantedin the user's brain. The electrical signal is sent to the electrodearray 114, causing one or more electrodes located on the array todeliver an electrical stimulus to a portion of the user's brain. Thestimulus causes the user to perceive at least a characteristic of thesound.

FIG. 4 is a functional block diagram illustrating a model 400 of inputsand outputs that may be utilized by a processor, such as the soundprocessor 208, to determine the dynamic range, according to variousembodiments. The model 400 includes a plurality of dynamic inputs 402, aplurality of user inputs 404, a plurality of static inputs 406, and anoutput 408.

The model 400 is preferably implemented as a set of computer-readableinstructions stored on a computer-readable medium configured to cause aprocessor to perform functions described by the instructions. Forexample, the instructions may be software or firmware stored on amemory, such as in the data storage 206. Alternatively, the model 400may be implemented partly or entirely by hardware, such as one or moreelectronics components. The model 400 preferably receives and storesinputs in a memory, such as the data storage 206, accessible by theprocessor. A user interface may be included to allow a user or device toprovide one or more of the inputs to the model 400. The outputpreferably consists of one or more signals and/or stored data values tobe used to provide stimulation. According to one example, the processoris the sound processor 208, the computer-readable medium is the datastorage 206, and the output is the transceiver 212 and inductive coil214.

The plurality of dynamic inputs 402 includes a signal input, a measureinput, and a noise estimate input. Each of these dynamic inputs 402 isupdated periodically, predictably, or sporadically. For example, one ormore of the dynamic inputs 402 could be updated for each sample of theinput audio signal received at the auditory device. The signal input ispreferably a channelized version of the input audio signal. The measureinput is preferably a statistical measure of upper bounds of the inputaudio signal, such as a percentile estimate, determined by a processingunit, such as the sound processor 208. For example, the measure inputcould be a 95^(th)-percentile estimate of the channelized input audiosignal. The noise estimate input is preferably performed with a process,such as within a preliminary noise reduction process. While theplurality of dynamic inputs 402 preferably comprises the above-describedthree inputs, more or fewer dynamic inputs may also be included in someembodiments.

The plurality of user inputs 404 includes a volume input and asensitivity input. Each of these user inputs 404 is updated on anasynchronous basis, but while the model is operating. The user inputs404 are variable parameters that the user can adjust to change listeningquality. For example, the volume input adjusts a maximum volume to beassigned to a maximum (or near-maximum) stimulation experienced by theuser (i.e. an upper value of an input dynamic range). The sensitivityinput adjusts a minimum volume to be assigned to a minimum (ornear-minimum) stimulation experienced by the user (i.e. a lower value ofthe input dynamic range). While the plurality of user inputs 404preferably comprises the volume and sensitivity inputs, more or feweruser inputs may also be included in some embodiments. In someembodiments, there are no user inputs 404.

The plurality of static inputs 406 is a set of fixed parameters thatremain unchanged while the model is operating and should be set beforeoperation of the auditory device. For example, one or more of the staticinputs are user-specific (e.g. recipient-specific) parameters set by anaudiologist during fitting of the auditory device. As another example,one or more of the static inputs 406 are device-specific parametersrelated to the particular auditory device (e.g. make, model, and/orunique device).

The plurality of static inputs 406 includes a minimum sound pressurelevel (Min SPL) input, a maximum sound pressure level (Max SPL) input, aT-Level input, a C-Level input, and Q-Value input (also called aQ-Factor input herein). The Min SPL input is a device-specific parametercorresponding to the functional noise floor of the stimulation device.The Max SPL input is a device-specific parameter corresponding to amaximum value of an analog-to-digital converter in the stimulationdevice. The T-Level input is a user-specific parameter (e.g. set by anaudiologist) that serves as a minimum (or near minimum) of an electricaldynamic range of the acoustic-to-electric mapping function. The C-Levelinput is a user-specific parameter (e.g. set by an audiologist) thatserves as a maximum (or near maximum) of an electrical dynamic range ofthe acoustic-to-electric mapping function. The Q-Value input is an inputcorresponding to the shape of the mapping function. The Q-Value shouldbe fixed for even loudness spacing between current steps (if electricalstimulation is used) or should be adjusted to produce an appropriatecurve for a natural-sounding output signal for acoustic applications,such as hearing aid devices.

The output 408 of the model 400 is a functional representation of astimulus level to be applied to the user by a stimulation deviceassociated with the auditory device. Thus, a mapping according to themodel 400 includes determining a stimulus level according to one or moreof a stimulus level, current level, a pulse width, a vibration rate, andan interphase gap. For example, for electrical stimulation, the output408 may include information for use in determining a signal amplitude, apulse width, or an interface gap for the stimulation signal, or somecombination of these, for example. For acoustic or vibratorystimulation, the output 408 may represent a speaker's output volume oramplitude or vibratory output amplitude or volume, for example.

FIG. 5 is a functional block diagram illustrating a device 500 fordynamically adjusting an input dynamic range, in accordance withembodiments of the present invention. Such a device could be used inplace of typical, more complex, multi-stage solutions, with possiblyonly front-end beam forming and weak front-end noise reduction remainingfrom such solutions, for example. The device 500 includes a measuremodule 502, a noise estimate module 504, a rules module 506, a breakrules module 508, a smoothing module 510, a scaling function module 528having a scaling value input 530, and a mapping function module 512having as inputs a volume input 514, a sensitivity input 516, and aQ-Value input 518. The device 500 performs operations on an input signal520 to obtain an output signal 522. A flow-forward structure isillustrated for slow SPL adjustments, while a feedback structure isillustrated for fast CSPL changes (e.g. see the description accompanyingthe break rules module 508, below). The device 500 can be utilized on aper-channel or multi-channel (clustered) basis, as well as on FFT bins(e.g. for acoustic applications) or in a broadband implementation. Inpreferred embodiments, for example, the device 500 could operate using22 channels (electric) or 65 bins (acoustic).

The measure module 502 preferably takes in the input signal 520 (or,alternatively and as illustrated in FIG. 5, a multi-channel signal viaspectral analysis 2 module 524) and a break signal from the break rulesmodule 508 to determine (1) an amplitude data (e.g. a percentileestimate or other measure), (2) a time constant for the measure, and (3)an initial condition. If the measure is a percentile estimate, then themeasure module 502 calculates a percentile estimate (e.g. 95%), in whichthe smoothed output will have 95% of the input signal amplitude below itand 5% of the input signal amplitude above it. In essence, the measuremodule 502 is tracking the peaks in the signal when it calculates apercentile estimate. The percentile estimate may range from 70-100%, forexample. In another example, the percentile estimate may be 50%, whichmay be more advantageous for certain acoustic auditory devices. In someembodiments, several measure modules may be included in parallel withone another, each calculating a different selectable percentile estimate(e.g. 70%, 75%, 80%) that may be used to define a dynamic-range mappingcurve. In other embodiments, one or more measure modules may calculate adifferent percentile estimate for each spectral channel. As describedbelow with respect to the break rules module 508, under somecircumstances, the break signal is activated, which quickly changes themapping curve according to a pertinent break rule from the break rulesmodule 508. The output of the measure module 502 is preferably used toset a dynamic CSPL to be used for mapping the input audio signal to acorresponding output stimulation.

The noise estimate module 504 preferably takes in the input signal 520(or a multichannel signal 526) and provides a mean noise level estimateas its output. Any of a variety of suitable known noise estimationstrategies, such as those currently used in certain hearing prostheses,may be utilized to determine this mean noise level. One example would beto use a percentile estimate such as the 25th percentile estimate. Otherexamples of concepts relating to noise estimation strategies may befound in the following two articles, the contents of both of which areincorporated by reference herein: R. Martin, “Noise Power SpectralDensity Estimation Based on Optimal Smoothing and Minimum Statistics,”IEEE Trans. On Speech and Audio Processing, vol. 9, no. 5, pp. 504-512,July 2005, and I. Cohen, “Noise Spectrum Estimation in AdverseEnvironments: Improved Minima Controlled Recursive Averaging,” IEEETrans. On Speech and Audio Processing, vol. 11, no. 5, pp. 466-475,September 2003. The noise estimate module 504 preferably uses anadaptation time of around 1 second. The output of the noise estimatemodule 504 is preferably used to set a dynamic TSPL to be used formapping the input audio signal to a corresponding stimulation.

As noted above, both the measure module 502 and the noise estimatemodule 504 receive an input signal, which may be the whole input signal520, or multiple spectrally limited signals such as FFT bins or channels(via spectral analysis 1 module 524 and spectral analysis 2 module 526,respectively). In some embodiments, the number of spectral bands, and/orthe widths of spectral bands used for the measure and for the noiseestimate may not be the same. In such a case, the input signal measurewith the greatest number of spectral bins preferably should determinethe number of spectrally limited mapping functions. For instance, themeasure may be calculated on a broadband signal provided by spectralanalysis 1 module 524, while the noise estimate may be calculated for 22spectral channels provided by spectral analysis module 526. In thiscase, the measure would be used for each of the 22 spectral channelmapping functions. Where partially overlapping spectral bands existbetween measures, an average, weighted average, minimum, or maximumvalue may be used for each measure to determine the measure or noiseestimate for each mapping function.

The rules module 506 helps to ensure that the outputs of the measuremodule 502 and/or the noise estimate module 504 result in a viable inputdynamic range. In other words, the rules module 506 constrains theabsolute and relative inputs used for mapping in order to prevent theauditory device from entering extraneous states.

FIG. 6 is a functional block diagram illustrating the rules module 506in further detail, according to an example. As shown, two inputs to therules module 506 are a dynamic CSPL input 602 and a dynamic TSPL input604, which are the outputs from the measure module 502 and the noiseestimate module 504, respectively.

A first rule imposed by the rules module 506 is an information ruleapplied by an information rule module 606. The information ruleessentially limits the absolute values of the dynamic CSPL, bypreventing the dynamic CSPL from going above a maximum limit, which maybe approximately 90 dB, for example. This 90 dB limit is slightly belowa maximum limit for a typical analog-to-digital converter (given aspecific microphone and amplifier combination). In addition, theinformation rule limits the absolute values of the dynamic CSPL fromgoing below a minimum limit, which may be approximately 40 dB, forexample. One example circumstance in which the information rule islikely to be invoked is when there is silence. In such a case, thedynamic TSPL input 604 would tend toward the noise floor of the inputmicrophone and the dynamic CSPL input 602 would also decrease toward thesame value. However, by applying the information rule, the dynamic CSPLis constrained to the minimum limit.

A second rule imposed by the rules module 506 is a noise rule applied bya noise rule module 608. The noise rule is similar to the informationrule, and essentially prevents the noise estimate (dynamic TSPL) fromgoing as low as the noise floor of the input microphone plus a buffer toaccount for system noise, which may be 20 dB, for example. It alsoprevents the dynamic TSPL from going too high (e.g. more than around 60dB).

A third rule imposed by the rules module 506 is a stretch rule appliedby a stretch rule module 610. The stretch rule prevents the inputdynamic range from being too large, preferably by increasing the dynamicTSPL level to ensure that the input dynamic range does not exceed amaximum value, such as 50 dB. For example, in a listening environmenthaving loud speech with negligible environmental noise, the noise floor(dynamic TSPL) will be small and the dynamic CSPL will be large. Byincreasing the dynamic TSPL (i.e. maintaining the dynamic CSPL, inaccordance with the first rule, above), the stretch rule helps to ensurethat system noise and low level environmental noise are mapped out ofthe stimulation.

A fourth rule imposed by the rules module 506 is a squash rule appliedby a squash rule module 612. The squash rule prevents the input dynamicrange from being too small, preferably by increasing the dynamic CSPL toensure that the input dynamic range is larger than a minimum value, suchas 30 dB. For example, in a listening environment having constantenvironment noise, the noise estimate (dynamic TSPL) and dynamic CSPLwill tend to become quite similar. By increasing the dynamic CPSL (i.e.maintaining the dynamic TSPL, in accordance with the second rule,above), the squash rule helps to ensure that the noise is mapped to alower stimulation level.

While the rules module 506 preferably applies all four of the aboverules, in some embodiments, fewer or more rules may be applied. Therules are preferably applied in a cascade order so that the desiredoutcome is not undone by one or more previous rules. A preferred outcomeof the application of the above rules is that the minimum dynamic TSPLwill be approximately 25 dB, the maximum dynamic CSPL will beapproximately 90 dB, and the dynamic range will be between approximately30 dB and approximately 50 dB. The outputs of the rules module 506 are anew dynamic CSPL output 614 and a new dynamic TSPL output 616 that canbe subsequently smoothed and mapped.

In a preferred implementation, the upper value maximum limit isapproximately 90 dB, the upper value minimum limit is approximately 40dB, the lower value maximum limit is approximately 60 dB, the lowervalue minimum limit corresponds to a noise floor of a microphoneassociated with the auditory device, the maximum difference threshold isapproximately 50 dB, and the minimum difference threshold isapproximately 30 dB.

The break rules module 508 receives the audio input signal 520 and thecurrent CSPL level (from the smoothing module 510) as inputs and, ifnecessary, applies relatively fast changes to the CSPL. While themeasure module 502, noise estimate module 504, and smoothing module 510make relatively slow changes to the CSPL and/or TSPL, in order toimprove listening quality, the break rules module 508 overrides theseother modules when a fast gain change needs to be made. For example, ifthe audio input signal 520 (or a function of the audio input signal)exceeds the current dynamic CSPL level by a certain threshold amount,then the break rules module 508 causes the dynamic CSPL to be increasedon a fast time constant basis, overriding the slow time constants of thesystem.

There are at least four ways in which the break rules module 508 canrespond to an input audio signal that exceeds the current CSPL level bythe certain threshold amount. A first way would be to do nothing, whichcauses at least a portion of the input audio signal to be above theoutput CSPL, potentially making it uncomfortably loud, until the slowertime constants has changed the dynamic CSPL, lowering the output signal.A second way would be to quickly and temporarily increase the dynamicCSPL to a suitably high level, and then decrease it back to the dynamicCSPL (either quickly or slowly) after the exceeding input audio signalis no longer present. A third way would be to quickly increase thedynamic CSPL to a suitably high level by changing the measure and thesmoothing of the measure, which may also require increasing the dynamicTSPL to maintain a suitably-sized input dynamic range. Finally, a fourthway would be to increase the dynamic CSPL to an intermediate level (e.g.halfway) between the original dynamic CSPL and the level related to thenew input audio signal. Each of these has its advantages anddisadvantages, and the selection of a particular technique can be user-or audiologist-selectable, or it can be set in the device duringmanufacture or provisioning, for example.

The smoothing module 510 smooths changes in the dynamic CSPL and dynamicTSPL to mitigate abrupt mapping changes and in order to provide the userwith the ability to better hear relative loudness, while maintainingperformance. Where there are multiple spectral channels and thereforemultiple mapping functions, the smoothing time may also vary acrossspectral channels.

One way to do this is for the smoothing module 510 to introduce arelatively long time constant (e.g. 30 seconds for dynamic CSPL) tochanges in the dynamic CSPL and dynamic TSPL. Time constants may rangefrom 2-180 seconds, for example, to provide suitable adaptation times.In contrast, the time constants used in the measure module 502 and noiseestimate module 504 are preferably much shorter (on the order of 1second) and attempt to provide the best estimates possible. Thesmoothing module 510 has as its inputs a break input from the breakrules module 508 and the new dynamic CSPL input and new dynamic TSPLinput received from the rules module 506. As outputs, the smoothingmodule 510 outputs a smoothed dynamic CSPL output (provided as an inputto the break rules module 508 and to the mapping function module 512)and a smoothed dynamic TSPL output provided to the mapping functionmodule 512.

Thus, according to one example relating to dynamic CSPL, if an audioinput signal becomes on-average louder, the signal will be above dynamicCSPL (e.g. in the CSPL+10% region) and will be perceived as loud forabout 30 seconds (corresponding to the time constant of the smoothingmodule 510) before the smoothing module 510 causes the dynamic CSPLlevel to increase and make the loud signal comfortable again (i.e. atthe C-Level). Conversely, if an audio input signal becomes on-averagesofter, the signal will be below dynamic CSPL and be perceived as softfor about 30 seconds before the dynamic CSPL level decreases and makesthe soft signal comfortably loud (more easily hearable) again.

Similarly, for dynamic TSPL, the smoothing module 510 attempts to changethe noise floor more slowly (e.g. with a 5 second time constant) thanthe short time constant (e.g. 1 second) typically implicit in noiseestimation. This will help to improve comfort and performance.

The smoothing module 510 may alternatively or additionally utilize amoving average to accomplish smoothing. For example, if the system 500stores a history of past determined dynamic TSPL and dynamic CSPL levelsin a memory associated with the auditory device (such as in the datastorage 206), the smoothing module 510 can periodically calculate amoving average of those stored determined levels. In a furtherembodiment, in order to introduce a predictive aspect to the smoothing,an autoregressive moving average (ARMA) may be utilized by the smoothingmodule 510. In a further embodiment, a finite impulse response (FIR) orinfinite impulse response (IIR) filter may be used, where the desiredsmoothing is achieved by adjusting the given time constants of the FIRor IIR filter. In still other embodiments, the mapping function 512 mayalso make use of a moving average, FIR, or IIR filter instead of or inaddition to smoothing module 510. The use of a moving average, FIR, orIIR filter in smoothing may be particularly beneficial foracoustic-to-acoustic mappings.

In a preferred embodiment, adaptation times are controlled to providedesired performance outcomes. Slow adaptation of the noise floor (noiseestimate) and the measure (e.g. percentile estimate) should be used todetermine the dynamic TSPL and dynamic CSPL, and therefore, the dynamicrange. Small deviations (e.g. less than 10 dB, for example) for shortterm periods should be possible, such as a fast time constant change(e.g. from the break rule), to increase the dynamic CSPL temporarily, inorder to decrease the effect on longer-term dynamic CSPL levels from theuser's own voice. Table A, below, illustrates presently preferred timeconstants. Other time constants, such as those determined during benchtesting and/or normal hearing testing, may be more desirable forparticular applications. Again, in general, relatively long (slow)adaptation times are used for the noise estimate (dynamic TSPL) andmeasure (dynamic CSPL), while a short (fast) adaptation time is usedwhen a break rule applies.

TABLE A Rise/Fall times Short term Short term Long term Long term risefall rise fall TSPL NA NA  5 seconds 20 seconds CSPL 0.01 seconds 0.1seconds 20 seconds 60 seconds

The mapping function module 512 maps the important part of the signal(e.g. speech segments) using the electrical dynamic range. In apresently preferred embodiment, the mapping is a mostly linear mappingfrom SPL-to-current level (e.g. dynamic TSPL is mapped to an outputT-Level and dynamic CSPL is mapped to C-Level). To allow for someoverhead above the C-Level, dynamic CSPL may be mapped to a sub-maximalcomfort level (e.g. C-Level minus 10%) or the C-Level could becorrespondingly increased (e.g. C-Level+10%). In other embodiments, themapping may map input SPL to output SPL (see FIG. 8).

The mapping function module 512 receives as inputs the input audiosignal 520 to be mapped, the smoothed dynamic CSPL, and the smootheddynamic TSPL. Additional inputs may include a volume input 514, asensitivity input 516, and a Q-Value input 518. The volume input 514 andsensitivity input 516 are preferably set by the user to change a pointon the mapping curve (see FIGS. 7 and 8) relative to the dynamic CSPLand/or dynamic TSPL, while the Q-Value input 518 is preferably setinternally in the auditory device. An additional input that could alsobe included is a scaling value input 530 that mixes between the relativeloudness described above, which has dynamic TSPL and dynamic CSPLvalues, and an absolute loudness that has a fixed TSPL and CSPL mapping(e.g. at 25 dB and 65 dB, respectively). Such a scaling value input 530could be used to scale between relative (dynamic SPL values) andabsolute (fixed SPL values) loudness to suit user preference (viascaling function module 528). The scaling function module 528effectively scales between a fixed TSPL and a fixed CSPL (i.e. absoluteloudness) and the dynamic TSPL and dynamic CSPL described above, inaccordance with the value provided as the scaling value input 530. Forexample, a value of [0,0] at the scaling value input 530 could mean tooperate with fixed CSPL and fixed TSPL (i.e. dynamic range altering isturned off); a value of [0,1] could mean to operate with a fixed CSPLand a dynamic TSPL; a value of [0.5,0] could mean to operate with theaverage of the fixed and dynamic CSPL and a fixed TSPL; and a value of[1,1] could mean to operate with a dynamic CSPL and a dynamic TSPL. Themapping function module 512 outputs an output signal 522 representativeof the stimulus signal to be applied to the user. While the output isillustrated as being a current level value in acoustic-to-electricalapplications in the example of FIG. 7, the output could alternatively bean SPL output, for example, for acoustic-to-acoustic applications.

Similar to as described above with respect to the smoothing module 510,the mapping function 512 may include smoothing functionality, wherechanges to the output signal 522 from the mapping function 512 aresmoothed. For example, the mapping function 512 may include an FFT/iFFTalgorithm having a large overlap (e.g. 87.5%) to provide smoothing,which may be beneficial for acoustic-to-acoustic applications. In such acase, a shorter input/output buffer may be utilized, if lower latency isdesired.

In alternative embodiments, the mapping function module 512 uses to thetwo inputs from the scaling function module 528 (and other inputs, e.g.,the input from volume module 514) to calculate a gain applied to theinput audio signal 520 to generate the output signal 522.

The break rule 508 and adjustments to the input dynamic TSPL and inputdynamic CSPL could each operate independent of the operation of theother two. For example, a fixed-input TSPL of 25 dB could be used withthe break rules module 508 turned off (or omitted), but changes to theinput dynamic CSPL could still be made in accordance with theembodiments described herein. As another example, an input dynamic TSPLcould be used with the break rules module 508 turned on (or included),while keeping a fixed input CSPL, such as at 65 dB. Additional inputscould be specified to turn the break rules module 508 on or off. Ingeneral, it should be understood that each of the three above-mentionedcomponents, input TSPL, input dynamic CSPL, and break rules module 508is capable of operating independently of the other two components insome embodiments.

FIG. 7 is a graphical representation 700 of an acoustic-to-electricmapping curve 706 embodying concepts described herein. The mapping curve706 includes an acoustic x-axis 702 showing Sound Pressure Level (SPL)in dB and a corresponding electric y-axis 704 showing current level inCurrent Units (CU), where the units of current will vary depending onthe particular application or the electrode used (e.g. cochlear implant,brainstem implant, midbrain implant, etc.). If the particular auditorydevice provides stimulation to more than one channel (e.g. where eachchannel stimulates at a different frequency range) or bin, then aseparate mapping curve is preferably assigned to each such channel orbin.

As illustrated, the mapping curve 706 is a generally smooth curve having(1) a first approximately linear section (having a slope a1 x+b1) atlower SPL, (2) a second approximately linear section (having a slope a2x+b2, where a1>a2 and b1<b2) at lower SPL, and (3) a shoulder regionbetween the first and second approximately linear sections (near the95th percentile SPL in the example shown). The general shape of themapping curve 706 is typically chosen to approximate normal hearing orto accomplish a particular auditory goal associated with an auditorydevice's application. The Q-Factor (see the description accompanying themapping function module 512 with reference to FIG. 5, above) can bepre-programmed, set by the user (e.g. using the Q-Value input 518), orcalculated to satisfy a particular auditory goal. Varying the Q-Factorchanges the shape (e.g. slope) of the mapping curve 706; for example,from a more-bowed curve to a flatter curve, or vice versa.

For acoustic applications, a curve that is less linear or evennon-linear (e.g. exponential) might provide better performance.Adjustments to the Q-Factor will also likely be more applicable to anacoustic application, where changes to the shape of the mapping curve706 could be relatively more significant compared to electricalstimulation applications.

In accordance with embodiments of the present invention, and asillustrated in the method 900 of FIG. 9 for processing an audio signal,such as an input audio signal, the mapping curve 706 is set as follows.The method 900 includes determining an amplitude data, such as apercentile estimate, representing a characteristic of an amplitude ofthe audio signal (block 902). According to one example, determining theamplitude data includes the measure module 502 determining a measure ofa characteristic of an average peak amplitude of the audio signal, suchas a percentile estimate.

The method 900 further includes setting an input dynamic CSPL parameterbased on the determined amplitude measure (block 904). Using the exampleof FIG. 7, the input dynamic CSPL (representing an estimate of thesignal's 95th percentile) is on the x-axis 702 and is set to map to the“C-Level” on the y-axis 704).

The method 900 further includes setting an input dynamic TSPL parameter(representing an estimate of the noise floor) (block 906). Referring toFIG. 7, the input dynamic TSPL parameter shown on the x-axis 702 ismapped to the T-Level on the y-axis 704. In one example, setting theinput dynamic TSPL parameter includes determining a noise-level estimateduring a noise smoothing time period and setting the input dynamic TSPLparameter to the noise-level estimate, as described above with respectto the smoothing module 510 of FIG. 5.

The points corresponding to the C-Level and T-Level, along with theQ-Factor, help to define the mapping curve 706, according to someembodiments. Also, as described above with reference to FIG. 5, themeasure module 502, break rules module 508, and smoothing module 510 maybe applied to the input dynamic CSPL and the dynamic TSPL. For example,in some embodiments, the break rules module 508 applies one or more ofthe first through fourth break rules (or a different break rule) insetting the input dynamic CSPL parameter and/or input dynamic TSPLparameter, as described with reference to FIG. 6.

Once the input dynamic CSPL parameter and the input dynamic TSLPparameter are set, resulting in a corresponding adjustment to or settingof the mapping curve 706, the method 900 includes mapping the audiosignal to a stimulation signal corresponding to the audio signalaccording to the input dynamic CSPL parameter and the input dynamic TSPLparameter (block 908). This mapping is preferably constantlyrecalculated during operation of the system 500 by constantlydetermining new amplitude data (e.g. percentile estimates) and noiseestimates. The method 900 may further include providing a stimulation,such as an electrical or acoustic stimulation, according to thestimulation signal. For example, in one embodiment, the stimulationsignal is characterized by a predetermined C-Level current level and apredetermined T-Level current level, in which the input dynamic CSPLparameter is mapped to the C-Level current level and the input dynamicTSPL parameter is mapped to the T-Level current level.

FIG. 8 is a graphical representation 800 of an acoustic-to-acousticmapping curve 806 illustrating effects of adjustments to the additionaluser inputs for the mapping function module 512 described above withreference to FIG. 5. The acoustic-to-acoustic mapping curve 806 issimilar to the acoustic-to-electric mapping curve 706, and a descriptionof common elements between the two is not repeated here.

The mapping curve 806 includes a first point 808 corresponding to adynamic TSPL, a second point 810 corresponding to a dynamic CSPL, athird point 812 corresponding to a maximum CSPL, and a fourth point 814corresponding to an acoustic S-level.

The first point 808 represents an input dynamic TSPL parameter “TSPL”(Threshold Sound Pressure Level) that maps to a corresponding output SPLlevel (shown as output TSPL in FIG. 8). For example, according to oneembodiment, the input dynamic TSPL parameter is set by determining anoise estimate, such as a mean-noise-level estimate, over a noisesmoothing time period (e.g. 1 second) and setting the input dynamic TSPLparameter to that determined mean-noise-level estimate. The set dynamicTSPL parameter is mapped to the output TSPL level, which is preferablypredetermined to match characteristics of the auditory device (e.g. adevice-specific parameter) and the user (e.g. a user-specific (e.g.recipient-specific) parameter). For example, the output TSPL level maycorrespond to a point at which an associated stimulation at that outputTSPL level allows the user to just barely perceive an indication of theaudio signal.

The second point 810 represents an input dynamic CPSL parameter “CSPL”(Comfort Sound Pressure Level) that maps to a corresponding dynamic CSPLlevel (shown as “output CSPL” in FIG. 8). For example, according to oneembodiment, the input CSPL parameter is set by determining a measurerelating to an upper volume bounds, such as a percentile estimaterepresenting a percentile of the signal amplitude of an input audiosignal. The input dynamic CSPL parameter is set based on the determinedmeasure. The set input dynamic CSPL parameter is mapped to the outputCSPL level, which is preferably predetermined to match characteristicsof the auditory device (e.g. a device-specific parameter) and the user(e.g. a user-specific (or recipient-specific) parameter). For example,the output CSPL level may correspond to a point at which an associatedstimulation at levels above that output CSPL level becomesuncomfortable. The output TSPL and output CSPL serve as a respectiveminimum and comfortable level of an acoustic dynamic range of anacoustic-to-acoustic mapping function. Further modifications to theinput dynamic TSPL and dynamic CSPL (e.g. due to application of rulesand smoothing) may result in corresponding modifications to the dynamicrange.

The third point 812 represents a maximum parameter “Max CSPL” that mapsto a corresponding output level above the output CSPL level associatedwith the input dynamic CSPL parameter. In the example of FIG. 8, theinput maximum CSPL is mapped to an output CSPL level that is the outputCSPL+10%. While the output CSPL level is a “comfort” level, the levelabove the CSPL level will likely be uncomfortable (i.e. loud in volume)to the user, but not cause harm. However, since the output CSPL levelcan be set to a percentile estimate that is less than 100% of the inputsignal amplitude, the levels above the output CSPL level are not likelyto be long in duration. To the extent they are, then the input dynamicCSPL parameter will be accordingly adjusted to reflect the new higherpercentile estimate. Having an input dynamic range that allows forlevels above and below the output CSPL level for a time period enablesthe user to better perceive differences in relative volume.

The fourth point 814 represents an S parameter “input acoustic S-Level”that maps to a corresponding output acoustic S-level. This fourth point814 is preferably included to provide a map termination point for lowsound pressure levels.

The first, second, third, and fourth points 808-814 lie on a curve andhelp to define the mapping curve 806, as each of these points lies onthe mapping curve 806, according to a preferred embodiment. In addition,additional points on the mapping curve 806 may be introduced, such asone or more intermediate points between the first point 808 and thesecond point 810. For example, an additional percentile estimate mightbe introduced (e.g. at 75-percent) to better tailor the mapping curve806 to a particular application and to provide a low gradient just belowthe dynamic CSPL parameter to present a significant portion of the loudacoustic signal near the output CSPL level. Fewer than the four points808-814 may also be used, in some embodiments.

As shown in FIG. 8, the mapping curve 806 has three portions orsegments: (1) an acoustic-acoustic loudness portion 816, (2) a headportion 818, and (3) a tail portion 820. The acoustic-acoustic loudnessportion 816 includes the majority of the information in the input audiosignal to be conveyed to the user. The head portion 818 is directed tohigher-volume parts of the signal (i.e. those above the measure, such asthose having a larger amplitude than a particular percentile estimate).The tail portion 820 is directed to sound pressure levels below theinput dynamic TSPL parameter. As shown, the mapping curve 806 should begenerally smooth (linear or near-linear, perhaps with a slight curvatureto avoid any abrupt changes, such as near the CSPL, for example). Alsoas shown, acoustic-acoustic loudness portion 816 has an average slopegreater than that of the head portion 818 and less than that of the tailportion 820. If additional points, beyond the first through fourthpoints 808-814 are included (such as intermediate points between thefirst and second points 808 and 810), then the mapping curve 806 mayinclude more than three segments. The mapping curve 806 mayalternatively comprise fewer than three segments.

FIG. 8 also illustrates how additional inputs, such as user inputs andstatic inputs, can affect the mapping curve 806, according to someembodiments. For example, the volume input 514 is preferably a userinput that moves the second point 810 up or down in current level (oroutput SPL level), causing a corresponding perceived increase ordecrease in volume at a particular current level (or SPL). In effect,the volume input 514 is thus adjusting the maximum of the electricaldynamic range (or output acoustic dynamic range). The sensitivity input516 is preferably a user input that moves the first point 808 up or downin current level (or output acoustic dynamic range), causing acorresponding perceived increase or decrease in output current levels(or output SPL). The sensitivity input 516 is, therefore, used to adjustthe minimum current level (or SPL). The Q-Value input 518 is preferablya static input set internally in the auditory device to adjust theacoustic-electric loudness portion 816. As shown in FIG. 8, adjustingthe Q-Value input 518 causes the acoustic-electric loudness portion 816to exhibit more or less curvature (see the “Q-Factor” arrows) on thecurve 806. The Q-Factor may also be used to adjust the sections 818and/or 820 as well. The output of this figure is in SPL units, which canbe used for some implementations such as for acoustic or vibratoryoutputs. While the output is illustrated as being an SPL value (FIG. 8),in acoustic-to-electrical applications, for example, the output couldalternatively be a current level (using T-Level and C-Level for TSPL andCSPL, respectively).

FIG. 10 is a high-level flow diagram illustrating a method 1000 ofprocessing an audio signal, according to one embodiment and as describedin further detail above. A sound processor, such as the sound processor208 illustrated in FIG. 2, and further described with respect to FIGS. 4and 5, performs the steps of the method 1000. In block 1002, the method1000 includes determining a measure and a noise estimate. The measuremodule 502 and noise estimate module 504 of the device 500 of FIG. 5respectively determine the measure and noise estimate, according to oneembodiment. In one example, the method 1000 includes determining themeasure and/or the noise estimate for each of a plurality of frequencychannels in the audio signal, such as described with respect to thespectral analysis 1 module 524 and spectral analysis 2 module 526 ofFIG. 5.

In block 1004, the method 1000 includes adjusting an input dynamic rangebased on the measure and the noise estimate. In one example, the audiosignal is an input audio signal composed of a plurality of samples, andthe adjusting is performed for one of the plurality of samples. In afurther example in which the audio signal has a plurality of frequencychannels, the adjusting is performed for each channel in a channelizedaudio signal.

In block 1006, the method 1000 includes determining an output 1006representative of a stimulus level, based on the measure, the noiseestimate, and a static input, such as an asynchronously-received userinput corresponding to volume or sensitivity (see user inputs 404 inFIG. 4), for example. Alternatively, the static input includes adevice-specific parameter and a user-specific parameter (see staticinputs 406 in FIG. 4), such as an MIN-SPL input, a MAX-SPL input, aT-Levels input, a C-Levels input, and/or a Q-Factor input. In the casewhere the audio signal is a channelized input audio signal, the method1000 may include determining the output for each channel. In oneexample, determining the output may include determining a stimulus levelaccording to one or more of a stimulus level, current level, a pulsewidth, a vibration rate, and an interphase gap.

The method 1000 may further include communicating the output to astimulation device, such as the stimulation decoder 306 and stimulationcomponent 308 illustrated in FIG. 3. In one example, the stimulationdevice includes a first subcomponent that generates the stimulus basedon the output and sends the stimulus to a second component (e.g. anarray of cochlear electrodes) that delivers the stimulus to the bodypart (e.g. a cochlea) of the user.

While much of the description above is in terms of a cochlear implant,the methods are applicable to other auditory devices that are notcochlear implants. For instance, if the auditory device is a boneconduction device, the implanted unit includes a transducer instead ofthe electrode array. In this example, the implanted unit uses thetransducer to cause vibrations on the user skull capable of stimulatingthe user's cochlea. Similarly, for auditory devices other than hearingprostheses, there will likely be no implanted unit, and instead, adifferent type of stimulation device (e.g. speaker transducer orwearable vibratory device) will be used.

In the acoustic and vibratory context, as discussed above, non-linearmapping curves may be more appropriate. In addition, it may be desirableto limit the effective gain changes between neighboring channels (e.g.within about 6 dB of effective applied gain, produced by mapping, ofeach other). The primary difference, of course, will be in the type ofstimulation provided (vibrating a speaker or motor rather than applyinga stimulus signal to an electrode) and possibly in the domains used forcalculating the fitting curve (loudness growth function). However,application to the acoustic or vibratory context still entails alteringan input dynamic range during operation/use of an auditory device tohelp ensure that the most important part of an input audio signal isused for stimulation. At a high level, this is achieved by mapping anupper portion (e.g. a 95th percentile) of the input audio signal to anupper portion of the dynamic range and by mapping a noise level (e.g. anoise estimate) to a lower portion of the dynamic range.

The examples were illustrated in a unilateral cochlear implant system.However, some or all embodiments are also applicable to bilateralcochlear implant systems, as well as other systems incorporating morethan a signal auditory device. In the case of bilateral auditorydevices, each auditory device preferably includes a processing unit thatcommunicates (via a wired or wireless connection) information pertainingto the microphone signal and the algorithmic state of the contralateraldevice (e.g. the left side device communicates its microphone signal andalgorithmic state to the right side device, and vice versa).

A bilateral auditory device comprising a first processing unitassociated with a first device (such as a left-side device) and a secondprocessing unit associated with a second device (such as a right-sidedevice) may be configured to perform a method for processing an audiosignal according to variations of the methods described. For example, inone embodiment, the method includes the first processing unitdetermining an input dynamic range based, at least in part, on acommunication received from and associated with the second processingunit. Such a communication could include a dynamic TSPL, a dynamicsmoothed TSPL, a dynamic CSPL, a dynamic smoothed CSPL, a break ruleoutput, a scaling value input, a volume input, a sensitivity input,and/or a Q-Value input associated with the second processing unit, forexample. Upon determining the input dynamic range, the first processingunit could then determine an output representative of a stimulus levelfor the first device, based on the determined input dynamic range. Inanother embodiment, the first processing unit communicates thedetermined input dynamic range to the second processing unit. The secondprocessing unit may, in turn, similarly utilize the communicated dynamicrange or other communicated information in processing an audio signal.

Under such a bilateral auditory device system, the embodiments set forthherein preferably have access to the raw acoustic signal from both sidesof the auditory devices (either external to the body or implanted).Alternatively, if communication bandwidth is limited between the twosides, the embodiments preferably have access to the algorithmic stateof each side. For example, the two auditory devices may shareinformation on the signal-to-noise (SNR) ratio of each side and the gainand threshold states of each respective side's processing.

Bilateral communications permit the algorithm to make informed decisionson mapping adjustments such that important cues employed in binauralhearing are preserved in real-world listening environments. Suchpreservation of binaural cues includes inter-aural level differences andbinaural head shadow effects, thereby preserving the ability to localizethe target information (e.g. a speaker's voice) and mask competingnoises from the captured signals from each auditory device'smicrophones.

Bilateral communication also permits optimization and coordination ofthe front-end processing specific to the directionality of themicrophones, not excluding the application of beamforming technologies.Here, the knowledge of each sides' acoustic signal and/or state allowsfor the adjustment of the directivity to best match the algorithm'sbinaural state, such that one side may serve as a noise reference (e.g.for use in determining a noise estimate) and the other side may providetarget information content. A further benefit of this front-endoptimization is the adjustment of gain and threshold to ensure that eachside's respective loudness is matched in accordance with the currentlistening environment.

While various aspects and embodiments have been disclosed herein, otheraspects and embodiments will be apparent to those skilled in the art.The various aspects and embodiments disclosed herein are for purposes ofillustration and are not intended to be limiting, with the true scopebeing indicated by the following claims.

What is claimed is:
 1. A method of processing an audio signal at anauditory device, comprising: determining a measure and a noise estimateof the audio signal; adjusting, based on the measure and the noiseestimate, at least an upper portion of an input dynamic range of theauditory device; and determining an output representative of a stimuluslevel based on the measure, the noise estimate, and a static input. 2.The method of claim 1, further comprising communicating the output to astimulation device.
 3. The method of claim 1, wherein the audio signalis an input signal composed of a plurality of samples, and wherein theadjusting is performed for one of the plurality of samples.
 4. Themethod of claim 1, wherein determining the output includes applying atleast one user input corresponding to volume or sensitivity.
 5. Themethod of claim 4, further comprising asynchronously receiving the atleast one user input.
 6. The method of claim 1, wherein the static inputincludes at least one of device-specific parameter related to thespecific auditory device and a user-specific parameter set by anaudiologist during fitting of the auditory device to a user.
 7. Themethod of claim 1, wherein the audio signal is separated into aplurality of frequency channels, and wherein the method furthercomprises: determining at least one of the measure and the noiseestimate for an audio signal component in each of a plurality offrequency channels.
 8. The method of claim 7, further comprising:adjusting at least an upper portion of an input dynamic range of theauditory device for each of the frequency channels based on the measureand noise estimate associated with the audio signal component in arespective frequency channel; and determining an output for eachfrequency channel.
 9. The method of claim 1, further comprisingadjusting an acoustic-to-electric mapping function based on the measureand the noise estimate, wherein determining the output includes applyingthe acoustic-to-electric mapping function to the audio signal.
 10. Themethod of claim 1, wherein the static input includes a minimum soundpressure level, a maximum sound pressure level, a T-Level, and aC-Level, wherein the minimum sound pressure level corresponds to a noisefloor, wherein the maximum sound pressure level corresponds to a maximumvalue of an analog-to-digital conversion, and wherein the T-Level andC-Level are, respectively, a minimum and a maximum of an electricaldynamic range of an acoustic-to-electric mapping function determinedusing at least one of the measure, the noise estimate, and the staticinput.
 11. The method of claim 1, wherein the determining comprisescalculating a gain and adjusting the audio signal based on the gain. 12.The method of claim 1, wherein the measure is a statistical measure ofan upper amplitude of the audio signal over a preceding time period. 13.The method of claim 12, wherein the statistical measure is a percentileestimate.
 14. The method of claim 1, wherein the measure is a measure ofa characteristic of an average peak amplitude of the audio signal overthe preceding time period.
 15. The method of claim 1, wherein adjustingat least the upper portion of the input dynamic range based on themeasure and the noise estimate comprises: mapping the upper portion ofthe audio signal as identified by a measure of an upper amplitude of theaudio signal over the preceding time period to an input dynamic comfortsound pressure level (CSPL) parameter.
 16. The method of claim 15,further comprising: setting an input dynamic threshold sound pressurelevel (TSPL) parameter.
 17. The method of claim 16, further comprisingperforming at least one of the following determining and adjusting steps(a)-(f) prior to the determining an output representative of a stimuluslevel: (a) determining whether the input dynamic CSPL parameter isgreater than an upper value maximum limit, and if so, adjusting theinput dynamic CSPL parameter to no more than the upper value maximumlimit; (b) determining whether the input dynamic CSPL parameter is lessthan an upper value minimum limit, and if so, adjusting the inputdynamic CSPL parameter to no less than the upper value minimum limit;(c) determining whether the input dynamic TSPL parameter is greater thana lower value maximum limit, and if so, adjusting the input dynamic TSPLparameter to no more than the lower value maximum limit; (d) determiningwhether the input dynamic TSPL parameter is less than a lower valueminimum limit, and if so, adjusting the input dynamic TSPL parameter tono less than the lower value minimum limit; (e) determining whether adifference between the input dynamic CSPL parameter and the inputdynamic TSPL parameter is greater than a maximum difference threshold,and if so, adjusting at least one of the input dynamic CSPL parameterand the input dynamic TSPL parameter so that the difference is notgreater than the maximum difference threshold; and (f) determiningwhether the difference between the input dynamic CSPL parameter and theinput dynamic TSPL parameter is less than a minimum differencethreshold, and if so, adjusting at least one of the input dynamic CSPLparameter and the input dynamic TSPL parameter so that the difference isnot less than the minimum difference threshold.
 18. The method of claim16, wherein setting the input dynamic TSPL parameter comprises:determining a noise-level estimate during a noise smoothing time period;and setting the input dynamic TSPL parameter to the noise-levelestimate.
 19. The method of claim 15, further comprising: determiningwhether a magnitude of the audio signal exceeds the input dynamic CSPLparameter by a break threshold, and if so, increasing the input dynamicCSPL parameter.
 20. The method of claim 1, wherein the determining anoutput representative of a stimulus level includes: determining at leastone of a signal amplitude, a pulse width, and an interphase gap for astimulation signal.
 21. An auditory device, comprising: a processingunit configured to: determine a measure of an upper amplitude of theaudio signal over a preceding time period and a noise estimate of theaudio signal for at least one of a plurality of samples of an audiosignal; and identify, based on the noise estimate and the measure, aninput dynamic range for mapping the at least one of the plurality ofsamples of the audio signal to a corresponding stimulation signal. 22.The auditory device of claim 21, wherein the measure is a statisticalmeasure of an upper bound of the at least one of the plurality ofsamples of the audio signal over the preceding time period.
 23. Theauditory device of claim 22, wherein the statistical measure is apercentile estimate.
 24. The auditory device of claim 21, wherein themeasure is a measure of a characteristic of an average peak amplitude ofthe audio signal over the preceding time period.
 25. The auditory deviceof claim 21, wherein to identify the input dynamic range for mapping theat least one of the plurality of samples of the audio signal to acorresponding stimulation signal, the processing unit is configured tomap an upper portion of the at least one of the plurality of samples ofthe audio signal as identified by the measure to an input dynamiccomfort sound pressure level (CSPL) parameter.
 26. The auditory deviceof claim 21, wherein to identify, based on the noise estimate and themeasure, an input dynamic range for mapping the at least one of theplurality of samples of the audio signal to a corresponding stimulationsignal, the processor is configured to: adjust, based on at least themeasure, at least an upper portion of the input dynamic range.
 27. Theauditory device of claim 21, the device further comprising: astimulation unit for providing a stimulus generated based on thestimulation signal to a user of the auditory device, wherein thestimulus comprises one or more of an electrical stimulus and amechanical stimulus.
 28. The auditory device of claim 21, wherein theprocessing unit is further configurable to determine respective noiseestimates and measures for other of the plurality of samples and totemporally smooth a plurality of the noise estimates and a plurality ofthe measures while identifying the input dynamic range.
 29. The auditorydevice of claim 21, wherein the processing unit is further configurableto determine respective noise estimates and measures for other of theplurality of samples, to identify a plurality of input dynamic rangesbased on the determined respective noise estimates and measures, and totemporally smooth the identified plurality of input dynamic ranges formapping the audio signal to a corresponding stimulation signal.
 30. Theauditory device of claim 21, wherein the processing unit is configuredto apply at least one rule that uses at least one of the noise estimateand the measure to identify the input dynamic range, and wherein the atleast one rule comprises at least one of an information rule and a noiserule, wherein the information rule comprises determining at least one ofwhether an upper value of the input dynamic range is greater than anupper value maximum limit and whether the upper value of the inputdynamic range is less than an upper value minimum limit, and, upondetermining that the upper value of the input dynamic range is greaterthan the upper value maximum limit or less than the upper value minimumlimit, adjusting the upper value respectively to the upper value maximumlimit or the upper value minimum limit, and wherein the noise rulecomprises determining at least one of whether a lower value of the inputdynamic range is less than a lower value minimum limit and whether thelower value of the input dynamic range is greater than a lower valuemaximum limit, and, upon determining that the lower value of the inputdynamic range is less than the lower value minimum limit or greater thanthe lower value maximum limit, adjusting the lower value respectively tothe lower value minimum limit or the lower value maximum limit.
 31. Theauditory device of claim 21, wherein the processing unit is configuredto apply at least one rule that uses at least one of the noise estimateand the measure to identify the input dynamic range, and wherein the atleast one rule comprises at least one of a stretch rule and a squashrule, wherein the stretch rule comprises determining whether adifference between an upper value of the input dynamic range and a lowervalue of the input dynamic range is greater than a maximum differencethreshold, and if so, adjusting at least one of the upper value and thelower value so that the difference is not greater than the maximumdifference threshold, and wherein the squash rule comprises determiningwhether the difference between the upper value and the lower value isless than a minimum difference threshold, and if so, adjusting at leastone of the upper value and the lower value so that the difference is notless than the minimum difference threshold.
 32. The auditory device ofclaim 21, further comprising at least one user input to set at least oneof a volume and a sensitivity, wherein the volume is used in identifyingan upper value of at least one of the input dynamic range and an outputdynamic range, and wherein the sensitivity is used in identifying alower value of at least one of the input dynamic range and the outputdynamic range.
 33. The auditory device of claim 21, wherein theprocessing unit is further configurable to map the audio signal to thecorresponding stimulation signal according to a map using a lower valueof the input dynamic range and an upper value of the input dynamicrange, wherein the input dynamic range maps to a corresponding outputrange predetermined for the user and the stimulation device.
 34. Theauditory device of claim 21, wherein the at least one rule comprises abreak rule comprising determining whether a magnitude of the audiosignal exceeds an upper value of the input dynamic range by a breakthreshold, and if so, increasing the upper value of the input dynamicrange.